Method for controlling communication service in a telecommunication and communicator associated therewith

ABSTRACT

A mobile first terminal ( 1 ) has a call in progress with a second terminal ( 2 ) under a first communications service via a base station ( 10 ) of the access network of a first subsystem. A condition for transferring the call to a base station ( 20 ) of the radio access network of a second subsystem is detected in a radio network controller ( 11 ) of the first subsystem. A core network switch ( 12 ) which is linked to the radio network controller ( 11 ) of the first subsystem is informed of said detection of a call transfer condition. If the second subsystem cannot process the call under the first communications service, a service change is requested so that said call can continue under the second communications service.

TECHNICAL FIELD

The present invention relates to controlling circuit mode communicationsservices in a telecommunications system and more particularly tomonitoring communications services using different ways of coding speechin a heterogeneous telecommunications system.

BACKGROUND

In digital landline telephone systems, speech is digitized, encoded to astandard law used in public networks (in particular the A and μ laws),and conveyed by 64 kilo bits per second (kbps) circuits. In radiocommunications systems it is generally necessary to reduce this bitrate, in particular at the radio interface. To this end radio terminalsincorporate a speech compression function.

In some systems, such as the Global System for Mobile communications(GSM), which is a second generation (2G) radio communications system,speech is transported in compressed form in the radio access networkbetween base stations and a transcoder and rate adapter unit (TRAU)located between the radio access network and the core network, usuallyin 16 kbps pulse code modulation (PCM) channels, a PCM channelcorresponding on the radio interface to one full rate speech channel orto two half-rate channels, for example. The TRAU transcodes speechbetween the 64 kbps coding law A and the full rate or half-rate code.

Other systems, such as the Universal Mobile Telecommunications System(UMTS), which is a third generation (3G) radio communications system,offer circuit mode communication with an end-to-end bit rate of 64 kbps.This caters in particular for videotelephone calls, which require ahigher bit rate than telephone calls and therefore cannot be supportedadequately by second generation systems.

Speech coding/decoding for videotelephone calls at an end-to-end bitrate of 64 kbps, for example, may be carried out in accordance with theH.324 standard, which is described in ITU-T Recommendation H.324,including appendix C thereof (“Multimedia telephone terminals over errorprone channels”) and where appropriate appendix H thereof (“Mobilemultilink operation”), which covers this kind of coding/decoding.Technical Specification TS 26.111, version 5.0.0, “Codec forCircuit-switched Multimedia Telephony Service; Modifications to H.324”,published in June 2002 by the 3^(rd) Generation Partnership Project(3GPP), which is more specifically adapted to UMTS terminals, alsocovers this kind of coding/decoding.

Since it conveys voice and video simultaneously, a videotelephone callnecessitates a high transmission bit rate, which the UMTS can offer. Incontrast, the GSM generally proves unable to support videotelephonecalls since the maximum bit rate authorized on its radio segment is toolow for this purpose.

A problem arises in heterogeneous radio communications systems, forexample a system comprising second generation (GSM) plant and thirdgeneration (UMTS) plant, because a terminal may initiate avideotelephone call when it is under the control of 3G plant, which callcannot continue if the terminal is transferred (handed over) to thecontrol of 2G plant, for example as a result of the terminal moving toan area in which no 3G plant is available. The videotelephone call isthen cut off, which is particularly frustrating for the user.

SUMMARY

One object of the present invention is to limit the drawbacks referredto above by proposing at least partial call continuity in aheterogeneous system, for example continuity of its voice portion.

Another object of the invention is to propose a service change adaptedto retain the voice portion of a videotelephone call on transfer of thecall between radio communications equipments of different generations.

A further object of the invention is to propose a service change adaptedto retain only the voice portion of a videotelephone call on transfer ofthe call between radio communications equipments of differentgenerations without excessively degrading the quality of the voice callon transfer.

The invention therefore proposes a method of controlling communicationsservice in a telecommunications system comprising first and secondsubsystems each including a radio access network comprising basestations and a radio network controller connected to at least some ofsaid base stations and to a core network switch, the first subsystembeing adapted to support first and second communications services andthe second subsystem being adapted to support the second communicationsservice, the method comprising the following steps in the case of afirst mobile terminal having a call in progress with a second terminalunder the first communications service via a base station of the radioaccess network of the first subsystem:

-   -   the radio network controller of the first subsystem detecting a        call transfer condition for transferring the call to a base        station of the radio access network of the second subsystem;    -   informing the core network switch to which the radio network        controller of the first subsystem is connected of said detection        of a call transfer condition; and    -   if the second subsystem is not adapted to process the call under        the first communications service, requesting a service change in        order for said call to continue under the second communications        service.    -   Thus a service change is effected before executing the call        transfer to enable fallback ahead of time to a mode of operation        supported by the second subsystem.

The first subsystem may be a third generation subsystem and the secondsubsystem a second generation subsystem, for example, and the firstcommunications service may be a videotelephone service and the secondcommunications service a voice telephone service. In this case, thevideo component of the call will be stopped before transferring the callto the 2G subsystem and the audio component will be retained. Thisachieves some degree of call continuity, which could not have beenachieved if the service change had not been effected before executingthe call transfer.

A coding change advantageously accompanies the communications servicechange, the new form of coding being selected to enable use of thesecond communications service and to be supported by the secondsubsystem. For example, the coding associated with the firstcommunications service is compatible with the H.324 standard and thecoding associated with the second communications service is of theAdaptive MultiRate (AMR) type.

The inability of the second subsystem to process the call under thefirst communications service may be detected at the switch connected tothe radio network controller of the first subsystem followingtransmission of a request to transfer the call to a switch connected tothe radio network controller of the second subsystem and reception inresponse thereto of a transfer failure message.

On receiving a transfer failure message relayed from the switch to whichit is connected, the radio network controller of the first subsystemadvantageously attempts to initiate the call transfer procedure again,advising the switch of the continuing detection of a call transfercondition. A transfer failure message continues to be sent to the radionetwork controller until the service change procedure is completed. Whenthe service change procedure has been completed, the call can then betransferred in the normal way, the risk of failure being eliminated byvirtue of the prior service change in respect of the call.

The invention also proposes a core network switch of atelecommunications system comprising first and second subsystems eachincluding a radio access network comprising base stations and a radionetwork controller connected to at least some of said base stations, atleast some of the radio network controllers also being connected to saidcore network switch, the first subsystem being adapted to support firstand second communications services and the second subsystem beingadapted to support the second communications service, said core networkswitch comprising, in relation to a first mobile terminal having a callin progress with a second terminal under the first communicationsservice via a base station of the radio access network of the firstsubsystem:

-   -   means for receiving an indication that the radio network        controller of the first subsystem has detected a call transfer        condition for transferring the call to a base station of the        radio access network of the second subsystem; and    -   means for requesting a service change in order for said call to        continue under the second communications service if the second        subsystem is not adapted to process the call under the first        communications service.

Other features and advantages of the present invention emerge from thefollowing description of non-limiting embodiments of the invention,which is given with reference to the appended drawings, in which:

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a simplified diagram of one example of a telecommunicationssystem architecture in which the invention may be used;

FIG. 2 represents an exchange of signaling employed by the invention;and

FIG. 3 represents an exchange of signaling used in a service changeprocedure employed by the invention.

DETAILED DESCRIPTION

FIG. 1 is a diagram of a telecommunications system architecture in whichthe invention may be used. Radio terminals 1 and 2 are mobile terminalssupporting the UMTS third generation radio communications system. A callbetween these two terminals is advantageously effected via 3G plant, atleast some of which forms a subsystem of the telecommunications systemthat comprises a radio access network with a 3G base station 10 withwhich the terminal 1 communicates over a radio interface and a radionetwork controller (RNC) 11 controlling the base station 10 and the callinvolving the mobile terminal 1. The RNC 11 is also connected to a 3Gmobile-service switching centre (MSC) 12 of the core network.

A call is in progress between a mobile terminal 1 and a terminal 2. Amobile terminal is also referred to as a user equipment (UE). If theterminal 2 is also a mobile terminal (denoted UE 2), the call is routedvia other 3G plant at the UE 2 end, such as an MSC 13 and an RNC 14connected to the MSC 13 and controlling a 3G base station 15 to which UE2 is connected.

The following description refers by way of example to a point-to-pointcall between UE 1 and 2, although it is equally feasible for a call toinvolve a greater number of terminals, for example UE 1 and two or moreother terminals. The call referred to below is effected in circuit mode,meaning that a circuit is reserved for exchanges between the UE via 3Gplant.

Moreover, and as indicated in the introduction, UE 1 and 2 have aprotocol stack that supports coding of the call frames exchanged. Theform of coding used may depend on the required bit rate and a requiredquality of service. It may equally depend on the type of service to beprovided during the call. If the communications service envisaged is avideotelephone service, i.e. a service involving simultaneoustransmission of video and audio, the form of coding selected must allowfor transmission at a high bit rate to authorize a large quantity ofinformation to be conveyed fast. In contrast, if the service is a voicetelephone service, a form of coding that provides transmission at alower bit rate may be selected.

One example of a protocol suite enabling call coding that may be used bythe UE to communicate via the UMTS is the H.324 standard conforming toITU-T Recommendation H.324, including appendix C thereof (“Multimediatelephone terminals over error prone channels”) and where appropriateappendix H thereof (“Mobile multilink operation”) or, to be morespecific, the standard defined in Technical Specification TS 26.111,version 5.0.0, “Codec for Circuit-switched Multimedia Telephony Service;Modifications to H.324”, published in June 2002 by the 3GPP. Saidstandard adapts the H.324 protocol suite to the UMTS. For simplicity,the term H.324 is used below to refer to the protocol suite applied tothe UMTS.

A videotelephone call is set up between UE 1 and UE 2 using theprocedures set out in section 5.3.6 of Technical Specification TS24.008, version 5.9.0, “Mobile radio interface Layer 3 specification;Core network protocols; Stage 3 (Release 5)”, published by the 3GPP inSeptember 2003. Accordingly, if UE 1 wishes to set up a videotelephonecall with UE 2, it sends a SETUP message to the MSC 12 (via the basestation 10 and the RNC 11). This SETUP message indicates that UE 1supports two modes of operation (“Bearer capability IE” (“BC IE”)), onefor a videotelephone service and the other for a voice telephoneservice. The SETUP message also contains a “Repeat indicator” parameterset to the value “support of service change and fallback”. This meansthat UE 1 is able to support a service change and in particular fallbackfrom a videotelephone service to a voice telephone service.

The MSC 12 then responds by sending UE 1 a CALL PROCEEDING messageincluding an indication of the capacity of UE 2. If both modes ofoperation (BC) are reported in the CALL PROCEEDING message, this amountsto accepting the types of communications service requested by UE 1,namely videotelephone or speech in this example. If only one BC isincluded in the CALL PROCEEDING message, only one of the twocommunications services supported by UE 1 can be used, for example avoice telephone service.

The same type of signaling is used between the MSC 13 and UE 2. Thesecall set-up procedures are also defined in detail in TechnicalSpecification TS 23.172, version 5.2.0, “Technical realization ofCircuit Switched (CS) multimedia service UDI/RDI fallback and servicemodification; Stage 2 (Release 5)”, published in September 2003 by the3GPP.

It is considered below that UE 1 has set up a circuit modevideotelephone call with UE 2 using the set-up procedures describedabove. This kind of call necessitates an end-to-end transmission bitrate of 64 kbps because the videotelephone service transmits a largeamount of data because of the simultaneous transmission of video andaudio. In this case, no transcoding is effected to reduce the usable bitrate between UE 1 and 2, in particular in the core network of thetelecommunications system. A call of this kind is supported by the 3Gsubsystem represented in the upper portion of FIG. 1, as explainedabove.

UE 1 communicates with UE 2 in particular via the radio interfacebetween UE 1 and the base station 10. The RNC 11 controls the radioresources for UE 1 in particular, in accordance with the Radio ResourceControl (RRC) protocol defined in Technical Specification TS 125 331,version 5.6.0, published in September 2003 by the 3GPP. In particular,the RNC 11 detects the occurrence of certain radio conditions on theradio link between UE 1 and the base station 10 in order to initiate aprocedure for transferring the call to other communications resources,known as a “handover” procedure.

Note that handover may be either “hard handover”, which instantaneouslyswitches the call from a first base station to a second, or “softhandover”, in which there is a more or less lengthy stage during whichthe mobile communicates simultaneously with both base stations. In asoft handover, the RNC 11 maintains and updates an active set of basestations with which UE 1 is communicating at any given time. Softhandover then consists in adding base stations to that active set and/orremoving them from it.

To execute the handover procedure, UE 1 and the base station 10 carryout radio measurements that include, for example, measuring uplink anddownlink field levels between UE 1 and the base station 10, field levelson downlinks from stations adjoining the base station 10, for examplethe base station 20, and other types of measurements, for examplemeasurements relating to uplink and downlink quality between UE 1 andthe base station 10 (see section 8.4 of TS 125 331 cited above).

An RNC conventionally initiates handover if the measurements reported toit by the UE and the base station concerned indicate that a transfercondition is satisfied, for example because the current link between theUE and the base station has either too low a field level or a qualitythat is deemed to be too low. The RNC also decides, on the basis of themeasurements reported to it, which base station is in a position to takeover the call after the transfer (or to be added to the active set inthe case of a soft handover).

In the FIG. 1 example, the RNC 11 detects a condition for transferringthe call from the base station 10, which is a 3G unit, to the basestation 20, which is a 2G unit. This situation may arise, for example,if UE 1 is initially in the vicinity of the base station 10 and thenmoves towards the base station. The signal level received from the basestation 10 then weakens and that received from the base station 20becomes dominant.

The base station 20 is a 2G unit. Transcoding is effected by atranscoder and rate adaptation unit (TRAU) 23 located between the basestation 20 and the 2G MSC 22, for example in the base station controller(BSC) 21 controlling the base station 20. This transcoding matches a 64kbps data stream such as exists in particular in the portion of the corenetwork starting at the 2G MSC 22 to a stream having a lower bit rate of16 kbps on the radio segment of the 2G subsystem so that it can betransported on a 16 kbps PCM channel. In particular, exchanges over theradio interface involving the base station 20 are effected at a usablebit rate of 9.6 kbps.

If the high speed circuit switched data (HSCSD) function is implementedin the network portion which includes the base station 20, this usablebit rate may be increased to 14.4 kbps, or even higher if a plurality oftime slots are used for the same call over the radio interface for agiven communication (a usable bit rate of up to 57.6 kbps can beachieved if four time slots are used).

The videotelephone call in the 3G subsystem has a bit rate of 64 kbps,as explained above. Transferring the call to the 2G subsystem whichincludes the base station 20 cannot be envisaged because the bit rateoffered cannot support the videotelephone call.

In practice, initiating handover after detecting a condition fortransferring the call from the base station 10 to the base station 20 isreflected in the sending of a request message on behalf of the RNC 11 tothe MSC 12 (see FIG. 2, “RELOCATION REQUIRED”). When the request hasbeen forwarded by the 3G MSC 12 (where applicable via an intermediary 3GMSC) to the 2G MSC 22 that is connected to the BSC 21 (“HANDOVER PREPREQ”), a failure message (“HANDOVER PREP FAILURE”) is sent back from the2G MSC 22 to the 3G MSC 12 (see Section 8.4 of technical specificationTS 129 002, version 5.7.0, “Mobile Application Part (MAP)specification”, published in September 2003 by the 3GPP). The MSC 12responds to the handover request sent by the RNC 11 with a failuremessage (FIG. 2, “RELOCATION PREP FAILURE”), forwarding the failuremessage that it has itself received, with the cause 29, i.e. “RELOCATIONFAILURE IN TARGET CN/RNC OR TARGET SYSTEM” (see section 8.6 of TechnicalSpecification TS 125 413, version 5.6.0, “UTRAN Iu Interface RadioAccess Network; Application Part (RANAP) signaling”, published inSeptember 2003 by the 3GPP).

Once failure of handover has been noted, for example on reception by the3G MSC 12 of a failure message from the 2G MSC 22 or on transmission tothe RNC 11 of a failure message “RELOCATION PREP FAILURE” by the 3G MSC12, the core network initiates a procedure for modifying thecommunications service to change from a videotelephone service to avoice telephone service.

This kind of service change involves a change of codec, i.e. a change ofthe coding/decoding mode used in the core network and by UE 1 and 2.Speech coding for a voice call advantageously accommodates a much lowertransmission bit rate than coding for a videotelephone call.

FIG. 3 shows an example of the signaling used in the service changeprocedure initiated by the MSC 12, following failure of the 3G to 2Ghandover requested by the RNC 11.

Thus the 3G MSC 12, which is at the UE 1 end, sends a command to changecodec to the 3G MSC 13, which is at the UE 2 end. This command (FIG. 3,“MODIFY CODEC”) includes the codec selected to continue the call. Inthis example, an adaptive multirate (AMR) codec is preferably used tocode the voice call. AMR codecs use eight different modes with bit ratesfrom 12.2 kbps to only 4.75 kbps. Coding is applied to 20 milliseconds(ms) speech frames corresponding to 160 samples at a sampling frequencyof 8000 samples/s. The algebraic code excited linear prediction coder(ACELP) coding scheme is used. Another example of an audio codec is theG.723.1 codec standardized by the ITU. This offers two bit rates: 5.3and 6.3 kbps. It codes speech and other audio signals into 30 ms frames.

The core network further includes media gateways (MGW) 16 and 17 whichexecute and monitor coding and decoding of streams passing through them.Streams may be coded and decoded differently in the two network portionson respective opposite sides of a MGW platform (see section 5.3 ofTechnical Specification TS 123 153, version 5.6.0, “Out of BandTranscoder Control, Stage 2” published in September 2003 by the 3GPP).This mode of operation provides transcoding between two types of codec.

After the command to change codec is sent from the MSC 12 to the MSC 13,signaling is exchanged between the MSC 12 and the MGW 16 to which it isconnected and between the MSC 13 and the MGW 17 to which it isconnected. This exchange of signaling (FIG. 3, “MODIFY BEARERCHARACTERISTICS”) renders the streams inactive during the service changeprocedure, so as to avoid that the MGW 16 and 17 generate error messagesas a result of any potential inconsistency between the coding used oneither side during this transitory period. On the occasion of thisexchange of signaling, the MGW 16 and 17 are informed of the new codecto be adopted for continuing the call, namely an AMR codec in the FIG. 3example. During this step, the communications medium (bearer)characteristics are modified in accordance with the envisaged servicechange.

The RNC 11 and 14 are then informed by the MSC 12 and 13, respectively,of the codec selected in the context of the change of codec procedure(FIG. 3, “DIRECT TRANSFER”). They then acknowledge acting on thatinformation (FIG. 3, “DIRECT TRANSFER COMPLETE”).

Note that the RNC 14 may reject a codec change request, for examplebecause the codec selected is not supported on the radio segmentextending from the RNC 14 to UE 2. In this case, the RNC 14 sends arejection message (“DIRECT TRANSFER [MODIFY REJECT (AMR)]”), bringingabout a new modification of the characteristics of the bearer negotiatedbetween the MSC 13 and the MGW 17, in order to revert to avideotelephone codec. A codec change failure message (“CODECMODIFICATION FAILURE”) is then sent from the MSC 13 to the MSC 12. Thevideotelephone codec is then reselected for the segment between UE 1 andthe MGW 16. Thus no service change is effected for the call in thiscase.

When the MSC 12 and 13 receive a “DIRECT TRANSFER COMPLETE” message,each MSC indicates to the corresponding RNC (11 or 14, respectively)that the radio access bearer (RAB) of the call must be modified to takeinto account the change of codec (FIG. 3, “RAB ASSIGN MODIFY”).Signaling is also exchanged over the Iu interface in the context ofmodifying the RAB, between the MGW 16 and the RNC 11, between the MGW 17and the RNC 14, and between the MGW 16 and 17. After this exchange overthe Iu interface, each RNC acknowledges the corresponding RABmodification indication (FIG. 3, “RAB ASSIGN MODIFY RSP”).

Thereafter, signaling is again exchanged between the MSC 12 and the MGW16 to which it is connected and between the MSC 13 and the MGW 17 towhich it is connected to render the streams active again (FIG. 3,“MODIFY BEARER CHARACTERISTICS”).

Finally, on completion of the above steps, the MSC 13 sends the MSC 12an acknowledgement message to confirm the successful change of codec(FIG. 3, “SUCCESSFUL CODEC MODIFICATION”). This message announces theend of the service change procedure. In the examples shown in FIGS. 2and 3, the call between UE 1 and UE 2, which was initially avideotelephone call, is then reduced to a voice call using an AMR audiocodec over the network segments between an MGW and the corresponding UE.The voice call coded in this way then has a bit rate from 12.2 kbps toonly 4.75 kbps, depending on the AMR codec mode used. It is assumedbelow that the AMR coding used is supported by the 2G subsystem, i.e.that it provides a usable bit rate of less than 9.6 kbps at the radiointerface.

In an advantageous embodiment of the invention, the RNC 11 that hasreceived a message reporting failure of transfer of the call to a basestation of the radio access network of the 2G subsystem subsequentlysubmits a new call transfer request (FIG. 2, “RELOCATION REQUIRED”). TheMSC 12 then advantageously sends a new failure message to the RNC 11until the change of service procedure has been completed. Thus the RNC11 may submit many new call transfer requests.

When the change of service procedure has been completed, i.e. when thecall in progress between UE 1 and UE 2 has been switched from avideotelephone service to a voice only service, a new attempt to send acall transfer request from the RNC 11 to the MSC 12 then provides fortransferring the call from the 3G subsystem to the 2G subsystemrepresented in FIG. 1 (this is the standard 3G to 2G handoverprocedure).

On completion of this handover procedure, the call between UE 1 and UE 2is routed across the 2G subsystem shown in the lower portion of FIG. 1,i.e. via the 2G base station 20, the BSC 21 and the 2G MSC 22. The 3Gplant, such as the MSC 13, the RNC 14 and the base station 15 may alsocontinue to route the call to UE 2. If the 2G MSC 22 is not connecteddirectly to the 3G MSC 13, the call may then pass in transit through aset of MSC connecting these two MSC indirectly.

The handover procedure succeeds in this situation because a servicechange is effected beforehand, thus reducing the speech bit rate to avalue acceptable for the 2G subsystem, i.e. a bit rate at the radiointerface of less than 9.6 kbps. In particular, the same AMR codec isthen used at both ends of the transmission chain, thereby assuringconsistent speech coding/decoding. The voice component of the call isthen retained after the call is transferred.

Because the codec change was effected ahead of call transfer and as soonas a call transfer condition was detected, continuity of service for thevoice component of the call is achieved without significantly degradingits quality.

Note that if the call transfer represents a soft handover, the servicechange may be carried out in the manner described above as soon as a 2Gbase station (for example the base station 20) has been added to theactive set kept up to date for UE 1.

The converse service change (from voice to videotelephone) may becarried out in accordance with the invention if the call is transferredfrom a region in which all plant is of the second generation to a regionin which third generation plant is present. An appropriate codec is thenselected to replace the initial audio codec, in order to switch from thevoice only component of the call to the complete videotelephone call.

An advantageous embodiment of the invention makes good use of thebandwidth that is not used by the coded information transmitted over thecall circuit of the 2G subsystem. In this way, if the audio codec usedafter call transfer provides a bit rate below the 9.6 kbps maximumusable bit rate authorized over the radio portion of the GSM, forexample if a 4.75 kbps AMR codec is used, the remaining 4.85 kbps(9.6−4.75=4.85) may be used to transmit data in addition to the voicecall.

The invention claimed is:
 1. A method of controlling communications service in a telecommunications system comprising first and second subsystems, the first subsystem being adapted to support first and second communications services and the second subsystem being adapted to support the second communications service, the method comprising the following steps in the case of a first mobile terminal having a call in progress with a second terminal under the first communications service via the first subsystem: detecting a call transfer condition for transferring the call to the second subsystem; if the second subsystem is not adapted to process the call under the first communications service, changing service from the first communications service to the second communications service while the call is on the first subsystem and continuing the call using only the second service, where changing service further comprises changing codecs from a first codec supporting the first service to a second codec supporting the second service; and after the change of service is complete, transferring the call to the second subsystem.
 2. The method according to claim 1, wherein a radio network controller of the first subsystem is connected to a core network switch and a radio network controller of the second subsystem is connected to a second core network switch, wherein, after the first switch has been informed of said detection of a call transfer condition, a request to transfer the call from the first switch to the second switch is transmitted, and wherein the inability of the second subsystem to process the call under the first communications service is indicated to the first switch by a transfer failure message sent in response to said transmission of the call transfer request.
 3. The method according to claim 2 wherein said transfer failure message is sent to the first core network switch and is forwarded to the radio network controller of the first subsystem and the step of informing the first switch of detection by the radio network controller of the first subsystem of a call transfer condition for transferring the call to a base station of the radio access network of the second subsystem is repeated for as long as a transfer failure message is forwarded to the radio network controller of the first subsystem.
 4. The method according to claim 1, wherein the first subsystem is of the third generation and the second subsystem is of the second generation.
 5. The method according to claim 1, wherein the first communications service necessitates a higher transmission bit rate than the second communications service.
 6. The method according to claim 1, wherein each communications service is associated with coding over at least a segment of the call and the service change request includes a request to change the coding over said call segment.
 7. The method according to claim 6, wherein the coding associated with the first communications service is compatible with the H.324 standard.
 8. The method according to claim 6, wherein the first communications service is a videotelephone service.
 9. The method according to claim 1, wherein the second communications service is a voice telephone service.
 10. The method according to claim 9, wherein Adaptive Multirate (AMR) coding is associated with the second communications service.
 11. The method according to claim 1, wherein, if the second communications service necessitates a bit rate over a radio segment that is strictly lower than a maximum bit rate value authorized by the second subsystem, the surplus bit rate is used to transmit data via at least said base station of the radio access network of the second subsystem.
 12. The method according to claim 1, wherein the service change request is transmitted to the first mobile terminal and to the second terminal.
 13. The method according to claim 12, wherein the service change request is transmitted to the second terminal via at least a switch, a radio network controller and a base station to which the second terminal is connected.
 14. The method according to claim 1, wherein the service change request includes a request for modification of radio access bearer characteristics of the call respectively at the mobile first terminal end and at the second terminal end; and a change from a first codec to a second codec is affected before the call is transferred, where the first codec performs coding and decoding for the first and second communications services, and the second codec performs coding and decoding for the second communications service.
 15. A core network switch of a telecommunications system comprising first and second subsystems each including a radio access network comprising base stations and at least a radio network controller connected to at least some of said base stations, at least some of the radio network controllers also being connected to said core network switch, the first subsystem being adapted to support first and second communications services and the second subsystem being adapted to support the second communications service, said core network switch comprising, in relation to a first mobile terminal having a call in progress with a second terminal under the first communications service via a base station of the radio access network of the first subsystem: means for receiving an indication that the radio network controller of the first subsystem has detected a call transfer condition for transferring the call to a base station of the radio access network of the second subsystem; means for requesting a service change in order for said call to continue under the second communications service if the second subsystem is not adapted to process the call under the first communications service, changing service from the first communications service to the second communications service while the call is on the first subsystem and continuing the call using only the second service, where changing service further comprises changing codecs from a first codec supporting the first service to a second codec supporting the second service; and after the change of service is complete, transferring the call to the second subsystem.
 16. The switch according to claim 15, wherein the radio network controller of the first subsystem is connected to said core network switch and the radio network controller of the second subsystem is connected to a second core network switch, the switch further comprising means responding to reception of an indication that a call transfer condition has been detected by transmitting a call transfer request to the second switch and means for deducing that the second subsystem is not able to process the call under the first communications service from the reception of a transfer failure message in response to transmission of said call transfer request.
 17. The switch according to claim 16, further comprising means for forwarding said transfer failure message to the radio network controller of the first subsystem.
 18. The switch according to claim 15, wherein the first subsystem is of the third generation and the second subsystem is of the second generation.
 19. The switch according to claim 15, wherein the first communications service necessitates a higher transmission bit rate than the second communications service.
 20. The switch according to claim 15, wherein each communications service is associated with coding over at least a segment of the call and the means for requesting a service change comprise means for requesting a coding change over said segment of the call.
 21. The switch according to claim 20, wherein the coding associated with the first communications service is compatible with the H.324 standard.
 22. The switch according to claim 15, wherein the first communications service is a videotelephone service, and wherein the second communications service is a voice telephone service.
 23. The switch according to claim 15, wherein Adaptive Multi Rate (AMR) coding is associated with the second communications service.
 24. The switch according to any claim 15, wherein the means for requesting a service change comprise means for transmitting a service change request to change from the first communications service to the second communications service to the mobile first terminal and to the second terminal.
 25. The switch according to claim 24, wherein the means for transmitting a service change request to the second terminal are provided by at least a switch, a radio network controller and a base station to which the second terminal is connected.
 26. The switch according to claim 15, wherein the means for requesting a service change include means for requesting a modification of characteristics of at least a radio access bearer of the call; and wherein a change from a first codec to a second codec is affected before the call is transferred to the second subsystem, where the first codec supports the first and second communications services, and the second codec supports the second communications services.
 27. A method comprising: controlling communications service in a telecommunications system comprising first and second subsystems, the first subsystem being adapted to support first and second communications services and the second subsystem being adapted to support the second communications service, wherein in the case of a first mobile terminal having a call in progress with a second terminal under the first communications service via the first subsystem, detecting a call transfer condition for transferring the call to the second subsystem; if the second subsystem is not adapted to process the call under the first communications service, changing service from the first communications service to the second communications service while the call is on the first subsystem and continuing the call using only the second service, where changing service further comprises changing codecs from a first codec supporting the first service to a second codec supporting the second service; and after the change of service is complete, transferring the call to the second subsystem. 